What Makes a Good RIAA or Line Stage?


Hi Doug,

In a currently running thread on a certain RIAA / Line stage beginning with the letter "E", some very provocative comments were made that are of a general nature.

I fear that this conversation will be lost on the many individuals who have soured on the direction which that particular thread has taken. For the purpose of future searches of this archive, those interested in the "E" thread can click this link.

For the rest of us who are interested in some of the meta concepts involved in RIAA and Line Level circuits, I've kicked this thread off - rather than to hijack that other one. In that thread, you (Doug) mused about the differences between your Alap and Dan's Rhea/Calypso:

... the Alaap has the best power supplies I've heard in any tube preamp. This is (in my admittedly unqualified opinion) a major reason why it outplayed Dan's Rhea/Calypso, which sounded starved at dynamic peaks by comparison.

Knowing only a bit more than you, Doug, I too would bet the farm on Nick's p-s design being "better", but know here that "better" is a very open ended term. I'd love to hear Nick's comments (or Jim Hagerman's - who surfs this forum) on this topic, so I'll instigate a bit with some thoughts of my own. Perhaps we can gain some insight.

----

Power supplies are a lot like automobile engines - you have two basic categories:

1. The low revving, high torque variety, characteristic of the American muscle car and espoused by many s-s designers in the world of audio.

2. The high revving, low torque variety characteristic of double overhead cam, 4 valves per cylinder - typically espoused by the single-ended / horn crowd.

Now, just as in autos, each architecture has its own particular advantage, and we truly have a continuum from one extreme to the other..

Large, high-capacitance supplies (category 1) tend to go on forever, but when they run out of gas, it's a sorry sight. Smaller capacitance supplies (category 2) recharge more quickly - being more responsive to musical transients, but will run out of steam during extended, peak demands.

In my humble opinion, your Alap convinced Dan to get out his checkbook in part because of the balance that Nick struck between these two competing goals (an elegant balance), but also because of a design philosophy that actually took music into account.

Too many engineers lose sight of music.

Take this as one man's opinion and nothing more, but when I opened the lid on the dual mono p-s chassis of my friend's Aesthetix Io, my eyes popped out. I could scarcely believe the site of all of those 12AX7 tubes serving as voltage regulators - each one of them having their own 3-pin regulators (e.g. LM317, etc.) to run their filaments.

Please understand that my mention of the Aesthetix is anecdotal, as there are quite a few designs highly regarded designs which embody this approach. It's not my intent to single them out, but is rather a data point in the matrix of my experience.

I was fairly much an electronics design newbie at the time, and I was still piecing my reality together - specifically that design challenges become exponentially more difficult when you introduce too many variables (parts). Another thing I was in the process of learning is that you can over-filter a power supply.

Too much "muscle" in a power supply (as with people), means too little grace, speed, and flexibility.

If I had the skill that Jim Hagerman, Nick Doshi, or John Atwood have, then my design goal would be the athletic equivalent of a Bruce Lee - nimble, lightning quick and unfazed by any musical passage you could throw at it.

In contrast, many of the designs from the big boys remind me of offensive linemen in the National Football League. They do fine with heavy loads, and that's about it.

One has to wonder why someone would complicate matters to such an extent. Surely, they consider the results to be worth it, and many people whom I like and respect consider the results of designs espousing this philosophy of complexity to be an effort that achieves musical goals.

I would be the last person to dictate tastes in hi-fi - other than ask them to focus on the following two considerations:

1. Does this component give me insight into the musical intent of the performer? Does it help me make more "sense" out of things?

2. Will this component help me to enjoy EVERY SINGLE ONE of my recordings, and not just my audiophile recordings?

All other considerations are about sound effects and not music.

Cheers,
Thom @ Galibier
128x128thom_at_galibier_design
WOW! A great thread with many designers I respect. The seam I've been mining all my audio DIY life is phono stages.

There is so much reward for the effort and of course, a little frustration on the way :-)

As most here will know, a lot of analog's perceived "issues" are actually present in the phono stage and quite often attributed to the mechanical aspects as Thom alludes to.

The overload aspects of the phono stage encompasses all aspects of the design and is where many designers, not so here though, look at the basic requirements and assume the signal levels are low and limited to the audio stage
only. Compared to power amps they are but I think many look at headline specs, say a MM input of 5mV and leave it at
that. However take that figure at 1k and then project that to 20k which is 20dB higher and then things look way different. Include the ability to handle HF transients and things look different again. I only know any of this
through trying stuff out and listening and it is illuminating to have people put a technical perspective on this i.e. some of this stuff can be measured.

However the thing I design for is hard to measure OR may be measured if only we knew what to look at. I design for
the ability of the music to communicate it's message to me. Consequently I don't measure anything. Mainly because I can't as I don't have the equipment nor could I afford it of sufficient accuracy AFAIK. Knowing (I think) the
accuracy required to measure RIAA EQ deviation, I find it hard to believe some of the figures being claimed in this
thread. A deviation of 0.1dB is I think 1% accurate. To measure this accurately requires test instrumentation to be
an order of magnitude more accurate so that's 0.1%. Is test equipment of this caliber being used to verify this? And
regularly checked against a test standard? As an amateur, I go for the "model it" approach aiming for the best fit
to the curve and then used better than 1% parts to hopefully get within the 1% window so within the 0.1dB deviation which could be called +/- 0.05dB. Is it? I have no idea but as I have got the model better and my design has got better, it would seem so.

Talking abou the EQ curve, there is the matter of the cutterhead rolloff or the 3.18uS turnover point. I used to
approach this as a lone point however as someone pointed out to me once, this turnover point affects the curve
waaaay lower down and so now I sim it as a LP filter that my inverse EQ feeds the sim with. Correcting for it's
effects lower down got me closer to a better phono stage. So I would question in some way, the effects of making it
switchable IF the other components in the EQ are not also altered at the same time.

I found it real interesting Johnothan's calcs re the effect of cart resonance and the frequencies it occurs at. It got
me thinking about its effects on not only the audio cct but the power supply. I personally don't like regulation (we
all have our prejudices) as I like to think the power supply should be as Nic Doshi says, as benign as possible. My experience is that this is really hard and when using regulation, for me at least, it has proven to be impossible.

With all this HF stuff going on, the concept that most regulators have a vice like grip on proceedings is a myth and their contribution to the circuit becomes nearly as great as the audio circuit itself. I try and avoid things like that ... well in my head anyway :-)

It will be no surpirse therefore that I favour hollow state. Another area that seems to be a poor cousin in the
power supply design is the heaters. IME, they have as much inluence over the sonics as the HT and so in mine, they
get as much filtering as the HT. It's not surprising that many commercial offerings kinda pass this by as it becomes
real expensive to do. I have no idea what anyone here does so it's not a criticism, just an observation of those
circuits I've seen.

I find it at odds with my experience that you can have a great design that makes good records sound good and also
bad resordings sound good. IME, a great design makes all records sound better and that's not done by introducing
"flavour" by being coloured but by making it more technically able. Bad recordings, to me anyhow, usually present more of a challenge and a better design is able to meet this challenge without causing the circuit to hold a white flag up and sound horrible. As the design gets better, good recordings sound even better, bad recordings can also benefit and rise above awful to enjoyable BUT the real essence of a good design is that all recordings sound more different. It is this aspect that drives me on. The better the design, the more of my record collection opens up to become enjoyed.

Some that used to be shocking I have discovered are real treats now. This is why I focus on the communication of the
music being the sole arbiter of goodness in the design and as such, the rest seems to come with it.

Another thing I wonder about is the concept of channel seperation. A cartridge is only so so in this regard. Once I used to do dual mono re the PSU but now I use a single supply as this seems to ground the musicians better and they seem to play together better. Its seems to be one of those HiFi vs music trade offs. I also wonder how you can get two supplies to be perfectly the same regarding noise and grounding and so therefore be exactly the same at all frequencies. As I can't see how to do this and the results of a single supply in my designs to be superiour, I wonder how much we should chase this notion of chqannel seperation.

Thanks Thom for kicking this thread off. I hope it reveals a bit more as it goes. I just wish I came to find it earlier.

regards,

Stephen
www.izzy-wizzy.com/audio
I can vouch for a number of StephenR's discoveries. He mentioned the sonic importance of heater supplies, and how they affect the sound quality just as much as the B+ supply. It may sound crazy, but he's right. In my linestage, I have tried different heater supply components and they affect the overall sound quality just like swapping components in the B+. Filter caps with the same value capacitance can sound quite different. The biggest surprise for me was trying a large choke filter in the heater supply. I used a .15H choke with a DCR of only 1 ohm, which is larger than most power amp transformers. The sound of the linestage is much improved with the choke in the heaters. Go figure. There's just no substitute for keeping an open mind and trying different circuits and different components to find the one that is "just right."

Dave

Dave
>>is frequency response at the phono stage is magnified by the time it reaches the speakers?<<

No it isn't. Deviations add up, but they don't multiply. A 1dB phono error will give you a 1dB shift in frequency response at the speaker.

How much is that? Well, take your listening position. Now move your head about 6 inches in any direction (up, down, front, back, left, right). That change you hear is probably more than 1dB.

>>A deviation of 0.1dB is I think 1% accurate<<

Yep. Here's a plot I made years ago showing what happens when the capacitors are off by +/-5%. You get peak errors of about +/-0.4dB. I'm sure the same sort of thing happens with resistor tolerance.

www.hagtech.com/images/accuracy.gif

>>Johnothan's calcs re the effect of cart resonance<<

He got it right. Resonances can be ultrasonic.

www.hagtech.com/loading.html

More info on RIAA at:

www.hagtech.com/pdf/riaa.pdf
www.hagtech.com/equalization.html
www.kabusa.com.riaa.htm

jh
Dear Bob: In a " perfect system with perfect audio devices that 1db deviation or any deviation that comes from the phono stage will be the same at the speaker output, but there is no perfect audio devices.

A signal that comes from a phono stage has to pass through: cables/connectors to the line stage, then in to the line stage, again to cables/connectors to the amplifier, then inside the amplifier and through the speaker cables: in all of these single " stages " the signal is suffering a degradation over the deviations that already has the signal that comes from the phono stage, when this signal goes out the speakers, IMHO, those deviations will almost be magnified but only with measurements about we could know for sure how much.

Regards and enjoy the music.
Raul.
>>3.18uS turnover point ... switchable ... other components altered at the same time<<

You are indeed correct. The 3.18us cannot simply be switched in and out. Adding the corner affects the other component values.

jh